Chord Electronics DAVE D/A processor

Fifteen years? Has it really been 15 years since I reviewed what was then the flagship D/A processor from English company Chord Electronics? In the July 2002 issue, here's how I summed up my review of the Chord DAC64: "While the Chord Electronics DAC64 is undoubtedly expensive, it is eye-poppingly gorgeous. . . . many listeners should find its silky-smooth highs seductive, as well as its slightly larger-than-life lows." How times and prices changethe "undoubtedly expensive" DAC64 cost only $3040! I did make a couple of criticisms of the DAC64 in my review, but according to Wes Phillips, in his August 2007 review of Chord's revised DAC64, "the Choral Blu [CD transport] and Choral DAC64 are, together, the CD player we music lovers have long prayed for"even if, five years after my own review, the DAC64's price had risen to $5000.

Then, in late 2015, at an event at Manhattan retailer Stereo Exchange to introduce the impressive little Chord Mojo portable D/A headphone amplifier (which I reviewed in our February 2016 issue), I saw an early production sample of the DAVE. The DAVEfor Digital [to] Analog Veritas [in] Extremis (Truth in Extreme)is said by its designer, Rob Watts, to be the highest-performance DAC to come from Chord, but at a price: it costs $10,588.

I made a mental note to put the Chord DAVE on my "must-review" list.

Description
Without its matching stand, the DAVE is housed in a relatively small but undoubtedly elegant rectangular enclosure with rounded sides that is superficially identical to that of the DAC64. Whereas the older DAC had a small, convex glass window in its top, the DAVE's top panel features a large, circular, four-color display set into it at an angle, and accompanied by an array of four inset spherical silver buttons surrounding a larger central button. Other than a recessed ¼" headphone jack at the bottom right of the front panel and a deeply recessed Chord logo on the front left of the top panel, that's all there is to see.

The rear panel features an array of digital input and analog output jacks, all unmarked save the right-channel unbalanced RCA jack, which has a red ring. Both balanced and single-ended outputs are provided, and the digital inputs include AES/EBU, USB2.0, two TosLink, and two coaxial S/PDIF on BNC jacks. There are also four digital-output BNCs. But what's inside the DAVE's elegant exterior?

Filter Technology
When Chord's Rob Watts visited my office in spring 2016, I asked him what his priorities had been in designing the DAVE. Chord's previous DACs had featured what was called the Watts Transient Aligned (WTA) reconstruction filter, which is said to minimize timing errors. I asked Watts what he meant by "Transient Aligned."

"Digital audio's Achilles' heel is the timing of transients. . . . Transients are very important for the brain's processing and how we perceive sound. Transients affect how we perceive pitch, timbre, and the positions of objects within the soundstage . . . very small timing errors have a very big subjective impact. The timing is reconstructed by the interpolation filter in the DAC and conventional DACs have timing uncertainty due to their limited processing. I used extensive listening tests to create the WTA filter, to simulate as closely as possible the results of an infinite-tap filter."

Watts explained that when digital audio data are created by sampling an analog signal, as long as those data are bandwidth-limited with zero output at half the sample rate, a sinc-function reconstruction filter with an infinite number of coefficients, or taps, will result in perfect reconstruction of the original waveform with perfectly defined transients. "But we can't have an infinite tap length, because we would be waiting an infinite length of time for the signal to fall out," he continued. "However, I did find that the filter algorithm makes a big difference to sound quality, so using an optimal filter allows the number of taps to be reduced to a practical number."

I asked him how many filter taps are "practical."

"If you have a conventional filter with 100 taps, you will recover some of the transient information," Watts replied. "A 100-tap filter gives you sufficiently good frequency-domain performance, but not in the time domain. . . . Every time you increase the number of taps, you improve the perception of pitch, timbre gets betterbright instruments sound brighter, dark instruments sound darkerthe starting and stopping of notes becomes easier to hear, the localization of sounds get better. There is less listening fatiguethe brain has to do less processing of the information presented to it to understand what's going on."

The digital filter in the discontinued DAC64 had 1024 taps; the WTA filter in Chord's still-available Hugo TT [$4795] has a tap length of 26,368. What is the tap length in the DAVE, I asked.

"The Xilinx FPGA [field programmable gate array chip] in DAVE is 10 times larger than the one used in the Hugo. . . . We've got 164,000 taps in DAVE's WTA filter, implemented in 166 DSP cores running in parallel; some of them are cores in the FPGA, some of them are custom cores using the FPGA fabric."

Did Watts use the same filter for PCM and DSD data, decimating the latter into high-resolution PCM?

"I managed to run two separate programs in the FPGA, one for PCM and one for the non-decimating DSD filter," he clarified. "My goal for DAVE was to keep the subjective timing improvement in Hugo, improve the noise-shaper performance, and, in the time domain, really get the transients more accurate, keep the noise-floor modulation and distortion very lowand we've got the budget to do much more advanced analog electronics. However, it is not just the tap length that matters. The filter also needs to be optimized. In Hugo, I went from a single-stage WTA filter to three stages. The first stage oversamples the data eight times; the second stage takes that to 16 times, and is followed by a linear interpolation filter to go to 2048Fs [2048 times the original sample rate]; then there are two low-pass filters. What I'd done [before Hugo], there was just a single interpolation filter, but that caused problems with noise-floor modulation and jitter sensitivity. In DAVE, by going from 16Fs to a 256Fs filter, that would recover the timing in a more efficient, more elegant waya more mathematically correct way of doing it. And when I got the 256Fs filter in, it sharpened up the transients and the whole presentation became much faster, became more neutral [compared with simply increasing tap length].

"To do a 256Fs FIR filter wasn't easy because you haven't got many cycles availableit used eight DSP cores. I've still got the linear interpolator filter to take it to 2048Fs, and then the two low-pass filters. What this all means is that inside the device, [even before being used to reconstruct the analog signal,] digital data at 2048Fs look much closer to the reconstructed analog signalvery tiny steps. The benefit of this is that, with 8Fs data, the steps are large and are much more susceptible to jitter.

"To turn those hi-rez 32-bit, 2048Fs data to analog, that's the function of the noise shaper. I use a noise shaper to reduce word length to 4 or 5-bit data [to present to a DAC using discrete components]. The design of the noise shaper was crucial, and as I had a lot more gates to play with than with Hugo, I could run the noise shaper at a much faster rate. My noise shaper is running at 104MHz compared with the typical 6MHz. The benefit of this fast rate is that noise shaping is an iterative processit constructs a low-frequency signal by running backward and forward at a very fast rate. If you run at a faster rate, you get much better accuracy in the audioband . . . soundstage depth gets a lot better."

Watts ended up with a 17th-order noise shaper (!) with 350dB dynamic range (!!) in the audioband, equivalent to 50 bits resolution (!!!). He designed his first pulse-array DAC, using flip-flops with a high but constant switching rate, in 1994; the DAVE, he said, "uses a 20-element pulse-array DAC in an FPGA. It's got a second-order analog noise shaper for the output stage, as DAVE's analog output stage needs to drive low-impedance headphones."

I was at first puzzled by the idea of an analog noise shaperuntil I realized that, as a first-order digital noise shaper comprises a feedback loop around a single-sample delay, a first-order analog noise shaper is simply a conventional feedback loop around an amplification stage. But . . . a second-order analog noise shaper?

It feels like Digital was refined, than more refined, then refined yet again. phew.

There are only a few 'remnant' that still cling to vinyl as their primary music storage system, it just seems so awkward and cumbersome. Why would a new music lover want Vinyl expense and square footage demands?, especially since MQA Dac prices start at $200!

Right now, today, I'm figuring that everyone in the Audio Business wants to be Bob Stuart.

I don't get the evangelical response it has received in certain corners. It's like someone told a tale of finding some golden ears in a secret place hidden by an audio angel and created the "Book of MQA" and the cult took off from there.

Meridian has always been an Engineering Company ( like 3M Corp. ), they've been at the forefront of Digital advancement. I've know these guys since the early 1980s, they're a no-bullshi... type group.

I don't think that I 'believe' in Meridian or MQA but rather I trust Meridian and MQA. The down-side being that they are upper-middle class. Still, a person can access MQA with a $200 Explorer2, $300 for an Audioquest Dragonfly RED or go full desktop with a Mytek for $2,000. ( we can't and won't be able to get a Schiit Bifrost MQA, too bad, I'm a Schiit fanboy )

Ortofon sent their Man to Analog Planet Headquarters to show their New & Improved $4,000 Phono Cartridge, the video is now on YouTube. Please notice the vastness of the Vinyl Record Collection used as the Backdrop, it's gigantic ! Vinyl guys are boxed into storage of Vinyl and the Fragile nature of the Gear needed for playback, not to mention the total lack of portability & incredible expense.

CD is copyable so it's profit possibility for Artists is less ( perhaps considerably less )

MQA/Tidal is Global in nature, as the World adopts Streaming, new issues will manifest themselves. Will this become another limiting factor?

My hope is to have access to all the recorded music that's out there, on a rental basis. I don't want to rely on my own digital storage & hard copy storage systems.

If I dump my vast CD collection and abandon my hard drive storage systems to rely on Tidal's rental collection will I be boxed in and risk Tidal going out of business? ( and ever increasing Monthly usage charges ).

Seems like there's always something to be worried about, like getting the Big C.

Chord's view -- "going for broke" on no. of taps to use in their devices -- is interesting.
Only a few weeks ago, Stereophile had a YouTube video interview with heads of mbl (German company). I think they said tap qty was more of an art and science -- a Goldilocks approach. If you go too high (as far as mbl was concerned) you can start harming the sound.

implying resolution close to 20 bits, which is state-of-the-art DAC performance

I thought 20-bit was obsolete? Stereophile said so on their Facebook page.

20-bit chips are obsolete as they become the limiting factor in resolution when used in real-word audio circuits, ending up with <20-bit resolution. The very best current D/A processors achieve between 20 and 21-bit resolution, limited primarily by the thermal noise of the resistors used.

All resistors produce thermal noise. The noise can be reduced by using lower-value resistors and correspondingly higher currents, but there is a practical limit.

One point I neglected to make in my earlier posting that in the specific example of a D/A processor using 20-bit DAC chips, that product's, in my opinion, less-than-optimal way of reducing the bit depth from 24 to 20 would result in a noise floor that would correlated with the signal. This is potentially more audible than a constant level of noise.

Yeah, if they had used, say, an ESS9038PRO dac, the performance would have just as good, if not better with its huge 140dB dynamic range.

The DAVEs IMD spec is indeed excellent. But you have tested other expensive dacs here with superb specs similar to this one. So how does the DAVE sonically compare with these other dacs, some using traditional dac chips?

If you say, "The very best D/A processors achieve between 20-21 bit resolution..."
And you state the Dave achieve "close to" this, and yet in Yggdrasil you make no mention of this,
How useful is all this work testing if this info is not laid out for the readers?
Why do I have to read the comments to realize these details?

So I have to "read between the lines", that the Dave get close, and the yggy is in league with "The very best D/A achieve between 20and 21 bit resolution"....

You focus on noise quite a bit..
To me, noise is not the issue.

The issue is that I have heard many top end dacs with a very unnatural quiet black background which I KNOW is not true to the source.
These dacs remove the background while cleaning up the noise floor, thus loosing information.

I would look again into some noise floor aspect to find out why.

So far, the Dave IMHO produce one of the most beautiful music I ever heard, as I hear no flaws in the source,
While the Yggdrasil still shows me the flaws in the source, with a slightly larger soundstag,
again IMHO.
Edit :
I just realized the effects described of black background are rom lower bit resolution (!)

This may be an oversight or my lack of understanding. But if you notice, the -90 dBFS 16-bit tone plot has a 1.5 mV peak-to-peak range. This seems unusually high relative to other plots I've seen here.

That seems to imply that 0 dBFS is about 33.5 Vrms or so. Doesn't this DAC clip at 12.35 Vrms? (assuming 8.75V is rms and "+3dB" mode is therefore 12.35 Vrms)

Wouldn't that create the illusion of more effective number of bits because the 0 dBFS signal power is much higher, relative to noise floor, than when referenced to a perhaps more standard 2.0 or 4.0 Vrms? I say illusion because of clipping, and because not many downstream equipment (for example, an amplifier) may accept a 33.5 Vrms range with out any issues.

This may be an oversight or my lack of understanding. But if you notice, the -90 dBFS 16-bit tone plot has a 1.5 mV peak-to-peak range. This seems unusually high relative to other plots I've seen here.

Yes it is. I set the Chord's volume control to its maximum for this test, even though that would have resulted in clipping with a full-scale signal, because I couldn't replicate the manufacturer's own measurement at lower levels.

ultrabike wrote:

Wouldn't that create the illusion of more effective number of bits because the 0 dBFS signal power is much higher, relative to noise floor, than when referenced to a perhaps more standard 2.0 or 4.0 Vrms?

With this test I was more concerned with looking at waveform symmetry with the undithered data.

With this test I was more concerned with looking at waveform symmetry with the undithered data.

i.e., the fact that I boosted the volume to impracticable levels is not relevant to the test results. On the other hand you say:

Quote:

I set the Chord's volume control to its maximum for this test, even though that would have resulted in clipping with a full-scale signal, because I couldn't replicate the manufacturer's own measurement at lower levels.

i.e., the fact that I boosted the volume to impracticable levels is critical to the test results, as it is only by so doing that I can avoid the conclusion that the manufacturer's claims are spurious. I am reminded of a well known expression involving the eating of cake.

Could I suggest that you provide the results at a standardised level that does not clip the full scale output, so as to allow a fair comparison with other products you have objectively tested?

For the yggydrasil review, you measured similar test waveform of undithered 1kHz sinewave...
yet at 24-bit data, and at extremely small levels of 200uV peak to peak!.
Then with Dave you change your test parameters because you cant get a good "manufacturer" result???!

This leads to pics that are thus misleading for the Dave, in comparison to all other Dacs reviews.

I rather to see a uniformity in your testing to be able to get a true comparison to other dacs, especially the yggydrasil.

As it stands member "ultrabike" sheds light to this discrepancy in measuring uniformity.

Therefore I would like to know, if you cannot achieve the performance the manufacturer states for the Dave,
What were your initial findings before you decided to "juiced up" the settings?

I already heard the Dave personally and have commented positive about it, so not knocking it, but, in fairness..
Curious members want to know.

I've read all the reviews of the new Chord DACs, and for some reason I feel the team at stereophile doesn't bring up the key reason of the TAPs and the USP of Chord. These DACS all have the fluidity and gentleness of true analog, but this is never mentioned. If you check the review of even the QBD76 on what hi-fi they did identify this very desirable quality. When I play Metallica on Vinyl, and then hear them on my Chord Dave and Chord Mojo, well these do play them with all the energy, and tunefulness (without sounding Harsh) of the Vinyl. Have a lot of respect for JA and the team at stereophile, but would be nice to get Dave reviewed by Fremer and see if it passes his "analog ears" test. (I'm pretty confident he will be pleased)